1. Field
The disclosed embodiments relate to noise cancellation and automatic volume adjustment for mobile audio devices.
2. Background Information
FIG. 1 (Prior Art) is a diagram of an active noise cancellation system that removes noise and echoes from speech information. A first microphone MIC1, referred to as the speech reference microphone, is placed close to the desired speech source. It picks up acoustic speech information from the user of the cellular telephone and converts it into an electrical speech signal 1. This speech signal 1 is contaminated with background noise. A second microphone MC2, referred to as the noise reference microphone, is placed close to the noise sources or far away from the speech source. It picks up acoustic noise, and converts it into an electrical noise reference signal 2. The noise reference signal 2 is assumed to be relatively free of the desired speech information as compared to the speech signal. Separation circuitry 3 uses the noise reference signal to cancel noise and to separate incoming signals into a speech signal 4 and a noise signal 5. The speech signal 4 is relatively free of noise. Echo cancellation involves employing adaptive filters to mimic echo paths. Cancellers 6 and 7 subtract the generated echo signals from the signals output by microphones MIC1 and MIC2. There are many such active noise cancellation techniques and circuits that are practiced in various technical fields.
Mobile communication devices (for example, cellular telephones) generally have small physical dimensions. These small dimensions limit the distance between the multiple microphones of an active noise cancellation system. As a consequence, the noise reference signal is generally not free of the desired speech information and noise cancellation performance is limited. Simple active noise filtering techniques tend to cancel some of the desired speech signal while leaving some of the noise uncancelled.
A more sophisticated noise reduction technique known as Blind Source Separation (BSS) has been used in digital hearing-aids. In such a BSS system, two microphones of a noise cancellation system are located in the ears of the hearing aid user. Accordingly, neither one of the two microphones can be used primarily for picking up noise. Both microphones pick up speech as well as noise to be cancelled. A temporal anti-Habbian learning algorithm is used to separate noise and speech information. For further information, see “Symmetric Adaptive Maximum Likelihood Estimation For Noise Cancellation And Signal Separation”, Electronics Letters, 33(17): 1437-1438 (1997), by M. Girolami; and “Blind Separation Of Convolved Sources Based On Information Maximization”, IEEE Workshop On Neural Networks for Signal Processing, Kyoto, Japan (1996), by K. Torkkola. Because performance of the BSS system generally relies on impulse response symmetry and proper placement of the microphones, additional signal processing may be applied.
FIG. 2 (Prior Art) is a diagram of a system employed in some cellular telephones. When using a cellular telephone, a cellular telephone user may be listening to audio that has softer passages as well as other relatively louder passages. If the user is listening to the audio in a noisy environment, then background noise may prevent the user from hearing the softer passages. If the overall electrical signal being supplied to the speaker were simply amplified, then the softer passages would be amplified so that the user could hear the softer passages, but the louder passages may then be amplified to the point that clipping occurs. Such clipping would introduce undesirable distortion into the sound emitted from the speaker. The loud sound could also damage the ear of the user. To prevent such undesirable clipping and distortion, the amplitude of signal 8 is tracked in the time domain. The signal is amplified by a gain that is a function of the input amplitude such that if the signal is weaker then the signal is amplified by larger gain values whereas if the signal is stronger then the signal is amplified by smaller gain values. The dynamic range of the overall signal is therefore compressed. This process may be referred to as “compression” or “Audio Dynamic Range Control” (ADRC) and occurs in ADRC block 10.
A process known as Automatic Volume Control (AVC) 11 is then applied to the output of ADRC block 10. The level of background noise is detected by microphone 12 and related circuitry 13. Under low background noise conditions, the compressed signal 9 need not be amplified by AVC block 11 and is supplied to the speaker largely unamplified. Under high background noise conditions, however, compressed signal 9 is substantially amplified by AVC block 11. Due to the compression performed by ADRC block 10, clipping of the high amplitude portions of the audio is reduced or eliminated. The above description of FIG. 2 is a simplification. See U.S. Pat. No. 6,766,176 for a more detailed description.